Fix sequence number used by asterisk for outgoing RTP packets
There was no audio for 3-way conference when sRTP is used. For 2-way calls frame->seqno is taken from DSP and is used by asterisk for the sequence number in RTP headers. However for 3-way conference the sequence number is generated by asterisk and it has to be greater than the previous value, otherwise libsrtp refuses to forward 'too old' RTP packets.
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